comparison projects/basic_sensor/render.cpp @ 13:6adb088196a7

Fixed ADC bug; added a simple passthrough test

 #include <rtdk.h>
 #include <cmath>
 
 float gPhase;
 float gInverseSampleRate;
-int gNumChannels;
+int gMatrixFramesPerAudioFrame;
 
 // These settings are carried over from main.cpp
 // Setting global variables is an alternative approach
 // to passing a structure to userData in initialise_render()
 

 					   int numMatrixFramesPerPeriod,
 					   int numAudioFramesPerPeriod,
 					   float matrixSampleRate, float audioSampleRate,
 					   void *userData)
 {
-	if(numMatrixFramesPerPeriod*2 != numAudioFramesPerPeriod) {
-		rt_printf("Error: this example needs the matrix enabled, running at half audio rate\n");
+	if(numMatrixFramesPerPeriod == 0 || numMatrixFramesPerPeriod > numAudioFramesPerPeriod) {
+		rt_printf("Error: this example needs the matrix enabled, with 4 or 8 channels\n");
 		return false;
 	}
 
-	gNumChannels = numAudioChannels;
+	gMatrixFramesPerAudioFrame = numAudioFramesPerPeriod / numMatrixFramesPerPeriod;
 	gInverseSampleRate = 1.0 / audioSampleRate;
 	gPhase = 0.0;
 
 	return true;
 }

 
 	// There are twice as many audio frames as matrix frames since audio sample rate
 	// is twice as high
 
 	for(int n = 0; n < numAudioFrames; n++) {
-		if(!(n % 2)) {
+		if(!(n % gMatrixFramesPerAudioFrame)) {
 			// Even audio samples: update frequency and amplitude from the matrix
-			frequency = map((float)analogRead(gSensorInputFrequency, n/2), 0, MATRIX_MAX, 100, 1000);
-			amplitude = (float)analogRead(gSensorInputAmplitude, n/2) / MATRIX_MAX;
+			frequency = map(analogRead(gSensorInputFrequency, n/gMatrixFramesPerAudioFrame), 0, MATRIX_MAX, 100, 1000);
+			amplitude = (float)analogRead(gSensorInputAmplitude, n/gMatrixFramesPerAudioFrame) / MATRIX_MAX;
 		}
 
 		float out = amplitude * sinf(gPhase);
 
-		for(int channel = 0; channel < gNumChannels; channel++)
-			audioOut[n * gNumChannels + channel] = out;
+		for(int channel = 0; channel < gNumAudioChannels; channel++)
+			audioOut[n * gNumAudioChannels + channel] = out;
 
 		gPhase += 2.0 * M_PI * frequency * gInverseSampleRate;
 		if(gPhase > 2.0 * M_PI)
 			gPhase -= 2.0 * M_PI;
 	}